1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
12
13 #include "api/audio/audio_frame.h"
14 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
15 #include "api/neteq/neteq.h"
16 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
17 #include "modules/audio_coding/neteq/default_neteq_factory.h"
18 #include "modules/audio_coding/neteq/tools/audio_loop.h"
19 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
20 #include "rtc_base/checks.h"
21 #include "system_wrappers/include/clock.h"
22 #include "test/testsupport/file_utils.h"
23
24 using webrtc::NetEq;
25 using webrtc::test::AudioLoop;
26 using webrtc::test::RtpGenerator;
27
28 namespace webrtc {
29 namespace test {
30
Run(int runtime_ms,int lossrate,double drift_factor)31 int64_t NetEqPerformanceTest::Run(int runtime_ms,
32 int lossrate,
33 double drift_factor) {
34 const std::string kInputFileName =
35 webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
36 const int kSampRateHz = 32000;
37 const std::string kDecoderName = "pcm16-swb32";
38 const int kPayloadType = 95;
39
40 // Initialize NetEq instance.
41 NetEq::Config config;
42 config.sample_rate_hz = kSampRateHz;
43 webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
44 auto audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
45 auto neteq =
46 DefaultNetEqFactory().CreateNetEq(config, audio_decoder_factory, clock);
47 // Register decoder in |neteq|.
48 if (!neteq->RegisterPayloadType(kPayloadType,
49 SdpAudioFormat("l16", kSampRateHz, 1)))
50 return -1;
51
52 // Set up AudioLoop object.
53 AudioLoop audio_loop;
54 const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop.
55 const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms.
56 if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
57 kInputBlockSizeSamples))
58 return -1;
59
60 int32_t time_now_ms = 0;
61
62 // Get first input packet.
63 RTPHeader rtp_header;
64 RtpGenerator rtp_gen(kSampRateHz / 1000);
65 // Start with positive drift first half of simulation.
66 rtp_gen.set_drift_factor(drift_factor);
67 bool drift_flipped = false;
68 int32_t packet_input_time_ms =
69 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
70 auto input_samples = audio_loop.GetNextBlock();
71 if (input_samples.empty())
72 exit(1);
73 uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
74 size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
75 input_samples.size(), input_payload);
76 RTC_CHECK_EQ(sizeof(input_payload), payload_len);
77
78 // Main loop.
79 int64_t start_time_ms = clock->TimeInMilliseconds();
80 AudioFrame out_frame;
81 while (time_now_ms < runtime_ms) {
82 while (packet_input_time_ms <= time_now_ms) {
83 // Drop every N packets, where N = FLAG_lossrate.
84 bool lost = false;
85 if (lossrate > 0) {
86 lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
87 }
88 if (!lost) {
89 // Insert packet.
90 int error = neteq->InsertPacket(rtp_header, input_payload);
91 if (error != NetEq::kOK)
92 return -1;
93 }
94
95 // Get next packet.
96 packet_input_time_ms = rtp_gen.GetRtpHeader(
97 kPayloadType, kInputBlockSizeSamples, &rtp_header);
98 input_samples = audio_loop.GetNextBlock();
99 if (input_samples.empty())
100 return -1;
101 payload_len = WebRtcPcm16b_Encode(input_samples.data(),
102 input_samples.size(), input_payload);
103 RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
104 }
105
106 // Get output audio, but don't do anything with it.
107 bool muted;
108 int error = neteq->GetAudio(&out_frame, &muted);
109 RTC_CHECK(!muted);
110 if (error != NetEq::kOK)
111 return -1;
112
113 RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
114
115 static const int kOutputBlockSizeMs = 10;
116 time_now_ms += kOutputBlockSizeMs;
117 if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
118 // Apply negative drift second half of simulation.
119 rtp_gen.set_drift_factor(-drift_factor);
120 drift_flipped = true;
121 }
122 }
123 int64_t end_time_ms = clock->TimeInMilliseconds();
124 return end_time_ms - start_time_ms;
125 }
126
127 } // namespace test
128 } // namespace webrtc
129